EP2165566A1 - Procédé et système permettant d'offrir une assistance auditive à un utilisateur - Google Patents

Procédé et système permettant d'offrir une assistance auditive à un utilisateur

Info

Publication number
EP2165566A1
EP2165566A1 EP07725080A EP07725080A EP2165566A1 EP 2165566 A1 EP2165566 A1 EP 2165566A1 EP 07725080 A EP07725080 A EP 07725080A EP 07725080 A EP07725080 A EP 07725080A EP 2165566 A1 EP2165566 A1 EP 2165566A1
Authority
EP
European Patent Office
Prior art keywords
audio signals
unit
voice
captured
microphone arrangement
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Withdrawn
Application number
EP07725080A
Other languages
German (de)
English (en)
Inventor
François Marquis
Benjamin Heldner
Fabian Nater
Giuseppina Biundo Lotito
Roman Arnet
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Sonova Holding AG
Original Assignee
Phonak AG
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Phonak AG filed Critical Phonak AG
Publication of EP2165566A1 publication Critical patent/EP2165566A1/fr
Withdrawn legal-status Critical Current

Links

Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/40—Arrangements for obtaining a desired directivity characteristic
    • H04R25/407—Circuits for combining signals of a plurality of transducers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/40—Arrangements for obtaining a desired directivity characteristic
    • H04R25/405—Arrangements for obtaining a desired directivity characteristic by combining a plurality of transducers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/55—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception using an external connection, either wireless or wired
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
    • H04R2225/41—Detection or adaptation of hearing aid parameters or programs to listening situation, e.g. pub, forest
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
    • H04R2225/43—Signal processing in hearing aids to enhance the speech intelligibility
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
    • H04R2225/61—Aspects relating to mechanical or electronic switches or control elements, e.g. functioning
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/20—Processing of the output signals of the acoustic transducers of an array for obtaining a desired directivity characteristic
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/55—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception using an external connection, either wireless or wired
    • H04R25/552—Binaural

Definitions

  • the present invention relates to a method for providing hearing assistance to a user; it also relates to a corresponding system.
  • the invention relates to a system comprising a microphone arrangement for capturing audio signals, audio signal processing means and means for stimulating the hearing of the user according to the processed audio signals.
  • One type of hearing assistance systems is represented by wireless systems, wherein the microphone arrangement is part of a transmission unit for transmitting the audio signals via a wireless audio link to a receiver unit comprising or being connected to the stimulating means.
  • the wireless audio link is an narrow band FM radio link.
  • the benefit of such systems is that sound captured by a remote microphone at the transmission unit can be presented at a much better SNR to user wearing the receiver unit at his ear(s).
  • the stimulating means is loudspeaker which is part of the receiver unit or is connected thereto.
  • Such systems are particularly helpful in teaching environments for normal-hearing children suffering from auditory processing disorders (APD), wherein the teacher's voice is captured by the microphone of the transmission unit, and the corresponding audio signals are transmitted to and are reproduced by the receiver unit worn by the child, so that the teacher's voice can be heard by the child at an enhanced level, in particular with respect to the background noise level prevailing in the classroom. It is well known that presentation of the teacher's voice at such enhanced level supports the child in listening to the teacher.
  • APD auditory processing disorders
  • the receiver unit is connected to or integrated into a hearing instrument, such as a hearing aid.
  • a hearing instrument such as a hearing aid.
  • the benefit of such systems is that the microphone of the hearing instrument can be supplemented or replaced by the remote microphone which produces audio signals which are transmitted wirelessly to the FM receiver and thus to the hearing instrument.
  • FM systems have been standard equipment for children with hearing loss in educational settings for many years. Their merit lies in the fact that a microphone placed a few inches from the mouth of a person speaking receives speech at a much higher level than one placed several feet away. This increase in speech level corresponds to an increase in signal-to-noise ratio (SNR) due to the direct wireless connection to the listener's amplification system.
  • SNR signal-to-noise ratio
  • the resulting improvements of signal level and SNR in the listener's ear are recognized as the primary benefits of FM radio systems, as hearing-impaired individuals are at a significant disadvantage when processing signals with a poor acoustical SNR.
  • FM+M the FM plus hearing instrument combination
  • FM+ENV the FM plus hearing instrument combination
  • This operating mode allows the listener to perceive the speaker's voice from the remote microphone with a good SNR while the integrated hearing instrument microphone allows to listener to also hear environmental sounds. This allows the user/listener to hear and monitor his own voice, as well as voices of other people or environmental noise, as long as the loudness balance between the FM signal and the signal coming from the hearing instrument microphone is properly adjusted.
  • FM advantage measures the relative loudness of signals when both the FM signal and the hearing instrument microphone are active at the same time.
  • FM advantage compares the levels of the FM signal and the local microphone signal when the speaker and the user of an FM system are spaced by a distance of two meters, hi this example, the voice of the speaker will travel 30 cm to the input of the FM microphone at a level of approximately 80 dB-SPL, whereas only about 65 dB-SPL will remain of this original signal after traveling the 2 m distance to the microphone in the hearing instrument.
  • the ASHA guidelines recommend that the FM signal should have a level 10 dB higher than the level of the hearing instrument's microphone signal at the output of the user's hearing instrument.
  • the relative gain i.e. the ratio of the gain applied to the audio signals produced by the FM microphone and the gain applied to the audio signals produced by the hearing instrument microphone
  • the relative gain has to be set to a fixed value in order to achieve e.g. the recommended FM advantage of 1OdB under the above- mentioned specific conditions.
  • the audio output of the FM receiver has been adjusted in such a way that the desired FM advantage is either fixed or programmable by a professional, so that during use of the system the FM advantage - and hence the gain ratio - is constant in the FM+M mode of the FM receiver.
  • EP 0 563 194 Bl relates to a hearing system comprising a remote microphone/transmitter unit, a receiver unit worn at the user's body and a hearing aid. There is a radio link between the remote unit and the receiver unit, and there is an inductive link between the receiver unit and the hearing aid.
  • the remote unit and the receiver unit each comprise a microphone, with the audio signals of theses two microphones being mixed in a mixer.
  • a variable threshold noise-gate or voice-operated circuit may be interposed between the microphone of the receiver unit and the mixer, which circuit is primarily to be used if the remote unit is in a line- input mode, i.e. the microphone of the receiver then is not used.
  • WO 97/21325 Al relates to a hearing system comprising a remote unit with a microphone and an FM transmitter and an FM receiver connected to a hearing aid equipped with a microphone.
  • the hearing aid can be operated in three modes, i.e. "hearing aid only", “FM only” or "FM+M".
  • the maximum loudness of the hearing aid microphone audio signal is reduced by a fixed value between 1 and 10 dB below the maximum loudness of the FM microphone audio signal, for example by 4dB.
  • Both the FM microphone and the hearing aid microphone may be provided with an automatic gain control (AGC) unit.
  • AGC automatic gain control
  • WO 2004/100607 Al relates to a hearing system comprising a remote microphone, an FM transmitter and left-and right-ear hearing aids, each connected with an FM receiver.
  • Each hearing aid is equipped with a microphone, with the audio signals from a remote microphone and the respective hearing aid microphone being mixed in the hearing aid.
  • One of the hearing aids may be provided with a digital signal processor which is capable of analyzing and detecting the presence of speech and noise in the input audio signal from the FM receiver and which activates a controlled inverter if the detected noise level exceeds a predetermined limit when compared to the detected level, so that in one of the two hearing aids the audio signal from the remote microphone is phase-inverted in order to improve the SNR.
  • WO 02/30153 Al relates to a hearing system comprising an FM receiver connected to a digital hearing aid, with the FM receiver comprising a digital output interface in order to increase the flexibility in signal treatment compared to the usual audio input parallel to the hearing aid microphone, whereby the signal level can easily be individually adjusted to fit the microphone input and, if needed, different frequency characteristics can be applied.
  • the signal level can easily be individually adjusted to fit the microphone input and, if needed, different frequency characteristics can be applied.
  • the signal level can easily be individually adjusted to fit the microphone input and, if needed, different frequency characteristics can be applied.
  • FM or inductive receivers are equipped with a squelch function by which the audio signal in the receiver is muted if the level of the demodulated audio signal is too low in order to avoid user's perception of excessive noise due a too low sound pressure level at the remote microphone or due to a large distance between the transmission unit and the receiver unit exceeding the reach of the FM link, see for example EP 0 671 818 Bl and EP 1 619 926 Al.
  • Contemporary digital hearing aids are capable of permanently performing a classification of the present auditory scene captured by the hearing aid microphones in order to select that hearing aid operation mode which is most appropriate for the determined present auditory scene. Examples of such hearing aids including auditory scene analysis can be found in US 2002/0037087, US 2002/0090098, WO 02/032208 and US 2002/0150264.
  • binaural hearing systems are available, wherein there is provided a usually wireless link between the right ear hearing aid and the left ear hearing aid for exchanging data and audio signals between the hearing aids for improving binaural perception of sound. Examples of such binaural systems can be found in EP 1 651 005 A2, US 2004/0037442 Al and
  • Hearing aids comprising an acoustic beam-former are described, for example, in EP 1 005 783 Bl, EP 1 269 576 Bl, EP 1 391 138 Bl, EP 1 303 166 A2 and WO 00/68703.
  • the direction of the formed acoustic beam is controlled by the measured direction of arrival (DOA) of the sound captured by the microphones.
  • the DOA can be estimated by comparing the audio signals captured by a plurality of spaced apart microphones, for example, by comparing the respective phases. If the microphones are directional microphones, the DOA may be calculated by forming level ratios of the audio signals, see, for example, WO 00/68703. With two microphones the DOA can be estimated in two dimensions, and with three microphones the DOA can be estimated in three dimensions.
  • EP l 303 166 A2 the audio signal processing is switched from an omnidirectional mode to a directional mode once the voice of a certain speaker has been recognized by identifying the speaker from a plurality of known speakers.
  • the DOA of the voice of the speaker is estimated and the result is used to set the beam former such that it points into this direction.
  • EP 1 320 281 A2 relates to a binaural hearing system comprising a beam former, which is controlled by the DOA determined separately for each of the left ear unit and the right ear unit, which each are provided with two spaced-apart microphones.
  • EP 1 691 574 A2 relates to a wireless system, wherein the transmission unit comprises two spaced-apart microphones, a beam former and a classification unit for controlling the gain applied in the receiver unit to the transmitted audio signals according to the presently prevailing auditory scene.
  • the classification unit generates control commands which are transmitted to the receiver unit via a common link together with the audio signals.
  • the receiver unit may be part of or connected to a hearing instrument.
  • the classification unit comprises a voice energy estimator and a surrounding noise level estimator in order to decide whether there is a voice close to the microphones or not, with the gain to be applied in the receiver unit being set accordingly.
  • the voice energy estimator uses the output signal of the beam former for determining the total energy contained in the voice spectrum.
  • this object is achieved by a method as defined in claim 1 and by a system as defined in claim 34, respectively.
  • the invention is beneficial in that, by taking into account both the estimated total energy contained in the voice spectrum of the audio signals and the estimated value of the direction of arrival of the audio signals when judging whether a voice is present close to the microphone arrangement, a high reliability of the detection of close voice can be achieved.
  • the audio signals are transmitted by a transmission unit via a wireless audio link to a receiver unit comprising a gain control unit, with the gain applied to the received audio signals being set according to the presence or lack of close voice, as judged from the captured audio signals.
  • the transmission unit comprises the microphone arrangement.
  • the receiver unit may comprise the stimulating means or it may be connected to integrated in a hearing instrument.
  • At least one of the microphones of the microphone arrangement is part of a right ear hearing instrument and at least one of the microphones of the microphone arrangement is part of a left ear hearing instrument, with the audio signals captured by the microphone of each of the hearing instruments being transmitted via a preferably wireless audio link to the respective other one of the hearing instruments.
  • Fig. 1 is a schematic view of the use of a first embodiment of a hearing assistance system according to the invention
  • Fig. 2 is a schematic view of the transmission unit of the system of Fig. 1;
  • Fig. 3 is a diagram showing the signal amplitude versus frequency of the common audio signal / data transmission channel of the system of Fig. 1 ;
  • Fig. 4 is a block diagram of the transmission unit of the system of Fig. 1 ;
  • Fig. 5 is a block diagram of the receiver unit of the system of Fig. 1 ;
  • Fig. 6 is a diagram showing an example of the gain set by the gain control unit versus time
  • Fig. 7 is a schematic view of the use of a second embodiment of a hearing assistance system according to the invention.
  • Fig. 8 is a block diagram of the receiver unit of the system of Fig. 7;
  • Fig. 9 shows schematically an example in which the receiver unit is connected to a separate audio input of a hearing instrument
  • Fig. 10 shows schematically an example in which the receiver unit is connected in parallel to the microphone arrangement of a hearing instrument
  • Fig. 1 1 is a block diagram of a voice activity detector (VAD) according to the invention suitable also for applications other than that of Fig. 4;
  • VAD voice activity detector
  • Fig. 12 is a schematic view of the use of a third embodiment of a hearing assistance system according to the invention.
  • Fig. 13 is a block diagram of one of the hearing instruments of Fig. 12. A first example of the invention is illustrated in Figs. 1 to 6.
  • Audio signals and control data are sent from the transmission unit 102 via radio link 107 to the receiver unit 103 worn by a user/listener 101.
  • background/surrounding noise 106 may be present which will be both captured by the microphone arrangement 26 of the transmission unit 102 and the ears of the user 101.
  • the speaker 100 will be a teacher and the user 101 will be a normal-hearing child suffering from APD, with background noise 106 being generated by other pupils.
  • Fig. 2 is a schematic view of the transmission unit 102 which, in addition to the microphone arrangement 26, comprises a digital signal processor 122 and an FM transmitter 120.
  • the lower part is used to transmit the audio signals (i.e. the first audio signals) resulting from the microphone arrangement 26, while the upper part is used for transmitting data from the FM transmitter
  • the data link established thereby can be used for transmitting control commands relating to the gain to be set by the receiver unit 103 from the transmission unit 102 to the receiver unit 103, and it also can be used for transmitting general information or commands to the receiver unit 103.
  • the internal architecture of the FM transmission unit 102 is schematically shown in Fig. 4. As already mentioned above, the spaced apart omnidirectional microphones Ml and M2 of the microphone arrangement 26 capture both the speaker's voice 105 and the surrounding noise
  • Ml is the front microphone and M2 is the rear microphone.
  • the microphones Ml and M2 together are associated to a beam- former algorithm and form a directional microphone arrangement 26 which, according to Fig. 1, is placed at a relatively short distance to the mouth of the speaker 100 in order to insure a good SNR at the audio source and also to allow the use of easy to implement and fast algorithms for voice detection as will be explained in the following.
  • the converted digital signals from the microphones Ml and M2 are supplied to the unit 111 which comprises a beam- former implemented by a classical beam-former algorithm and a 5 kHz low pass filter.
  • the first audio signals leaving the beam former unit 111 are supplied to a gain model unit 112 which mainly consists of an automatic gain control (AGC) for avoiding an overmodulation of the transmitted audio signals.
  • AGC automatic gain control
  • the output of a gain model unit 112 is supplied to an adder unit 113 which mixes the first audio signals, which are limited to a range of 100 Hz to 5 kHz due to the 5 kHz low pass filter in the unit 111, and data signals supplied from a unit 116 within a range from 5 kHz and 7 kHz.
  • the combined audio/data signals are converted to analog by a digital- to-analog converter 119 and then are supplied to the FM transmitter 120 which uses the neck- loop 121 as an FM radio antenna .
  • the transmission unit 102 comprises a classification unit 134 which includes units 114, 115, 116, 117, 118 and 219, as will be explained in detail in the following.
  • the unit 114 is a voice energy estimator unit which uses the output signal of the beam former unit 111 in order to compute the total energy contained in the voice spectrum with a fast attack time in the range of a few milliseconds, preferably not more than 10 milliseconds. By using such short attack time it is ensured that the system is able to react very fast when the speaker 100 begins to speak.
  • the output of the voice energy estimator unit 114 is provided to a voice judgement unit 115.
  • the input signals to the beam-former unit 111 i.e. the digitized audio signals captured by the microphones Ml and M2, respectively, are also supplied as input to a direction of arrival (DOA) estimator 219 which is provided for estimating, by comparing the audio signals captured by the microphone Ml and the audio signals captured by the microphone M2, the DOA value of the captured audio signals.
  • DOA value indicates the Direction of Arrival estimated with the phase differences in the audio band of the incoming signal captured by the microphones Ml and M2.
  • the output of the DOA estimator 219 i.e. the estimated DOA value, is provided to the voice judgement unit 115.
  • the voice judgement unit decides, depending on the signals provided by the voice energy estimator 114 and the DOA estimator 219, whether close voice, i.e. the speaker's voice, is present at the microphone arrangement 26 or not.
  • close voice i.e. the speaker's voice
  • the voice detection in the DOA estimator 219 and the voice energy estimator unit 114 is independent of the direct audio path, their outputs can be computed from filtered input signals which may be confined with regard to frequency ranges.
  • Appropriate frequency bands are defined DOA estimator 219 and the voice energy estimator unit 114 with regard to the directivity pattern of the microphones Ml, M2 and the beam-former unit 1 11, and the spectra of voice to be detected and/or the noise signals to be rejected. Thresholds must be adjusted accordingly.
  • the DOA estimator 219 and the voice energy estimator unit 1 14 use only frequencies below 1 kHz. Thereby it can be avoided, for example, that screech sounds generated by a teacher writing in on the blackboard are erroneously detected as the teacher's voice.
  • the unit 117 is a surrounding noise level estimator unit which uses the audio signal produced by the omnidirectional rear microphone M2 in order to estimate the surrounding noise level present at the microphone arrangement 26.
  • the surrounding noise level estimator unit 117 is active only if no close voice is presently detected by the voice judgement unit 115 (in case that close voice is detected by the voice judgement unit 115, the surrounding noise level estimator unit 117 is disabled by a corresponding signal from the voice judgment unit 115).
  • a very long time constant in the range of 10 seconds is applied by the surrounding noise level estimator unit 117.
  • the surrounding noise level estimator unit 117 measures and analyzes the total energy contained in the whole spectrum of the audio signal of the microphone M2 (usually the surrounding noise in a classroom is caused by the voices of other pupils in the classroom). The long time constant ensures that only the time-averaged surrounding noise is measured and analyzed, but not specific short noise events.
  • a hysteresis function and a level definition is then applied in the level definition unit 118, and the data provided by the level definition unit 118 is supplied to the unit 1 16 in which the data is encoded by a digital encoder/modulator and is transmitted continuously with a digital modulation having a spectrum a range between 5 kHz and 7 kHz. That kind of modulation allows only relatively low bit rates and is well adapted for transmitting slowly varying parameters like the surrounding noise level provided by the level definition unit 118.
  • the estimated surrounding noise level definition provided by the level definition unit 118 is also supplied to the voice judgement unit 115 in order to be used to adapt accordingly to it the threshold level for the close voice/no close voice decision made by the voice judgement unit 115 in order to maintain a good SNR for the voice detection.
  • a very fast DTMF (dual-tone multi-frequency) command is generated by a DTMF generator included in the unit 1 16.
  • DTMF generator uses frequencies in the range of 5 kHz to 7 kHz.
  • the benefit of such DTMF modulation is that the generation and the decoding of the commands are very fast, in the range of a few milliseconds. This feature is very important for being able to send a very fast
  • the command signals produced in the unit 116 are provided to the adder unit 113, as already mentioned above.
  • the units 109 to 1 19 all can be realized by the digital signal processor 122 of the transmission unit 102.
  • the receiver unit 103 is schematically shown in Fig. 5.
  • the audio signals produced by the microphone arrangement 26 and processed by the units 11 1 and 112 of transmission unit 102 and the command signals produced by the classification unit 134 of the transmission unit 102 are transmitted from the transmission unit 102 over the same FM radio channel to the receiver unit 103 where the FM radio signals are received by the antenna 123 and are demodulated in an FM radio receiver 124.
  • An audio signal low pass filter 125 operating at 5 kHz supplies the audio signals to an amplifier 126 from where the audio signals are supplied to a power audio amplifier 137 which further amplifies the audio signals for being supplied to the loudspeaker 136 which converts the audio signal into sound waves stimulation the user's hearing.
  • the power amplifier 137 is controlled by a manually operable volume control 135.
  • the output signal of the FM radio receiver 124 is also filtered by a high pass filter 127 operating at 5 kHz in order to extract the commands from the unit 116 contained in the FM radio signal.
  • a filtered signal is supplied to a unit 128 including a DTMF decoder and a digital demodulator/decoder in order to decode the command signals from the voice judgement unit 115 and the surrounding noise level definition unit 118.
  • the command signals decoded in the unit 128 are provided separately to a parameter update unit 129 in which the parameters of the commands are updated according to information stored in an EEPROM 130 of the receiver unit 103.
  • the audio signal amplifier 126 which is gain controlled.
  • the audio signal output of the amplifier 126 - and thus the sound pressure level at which the audio signals are reproduced by the loudspeaker 136 - can be controlled according to the result of the auditory scene analysis performed in the classification unit 134 in order to control the gain applied to the audio signals from the microphone arrangement 26 of the transmission unit 102 according to the present auditory scene category determined by the classification unit 134.
  • Fig. 6 illustrates an example of how the gain may be controlled according to the determined present auditory scene category.
  • the voice judgement unit 1 15 provides at its output for a parameter signal which may have two different values:
  • the gain of the microphone arrangement 26 is reduced relative to the gain of the microphone arrangement 26 during "voice ON". This ensures an optimum SNR of the sound signals present at the user's ear, since at that time no useful audio signal is present at the microphone arrangement 26 of the transmission unit 102, so that user 101 may perceive ambient sound signals (for example voice from his neighbor in the classroom) without disturbance by noise of the microphone arrangement 26.
  • the control data/command issued by the surrounding noise level definition unit 1 18 is the "surrounding noise level" which has a value according to the detected surrounding noise level.
  • the "surrounding noise level” is estimated only during “voice OFF” but the level values are sent continuously over the data link.
  • the parameter update unit 129 controls the amplifier 126 such that according to the definition stored in the EEPROM 130 the amplifier 126 applies an additional gain offset to the audio signals sent to the power amplifier 137.
  • the "surrounding noise level” is estimated only or also during "voice ON”.
  • the parameter update unit 129 controls the amplifier 126 depending on the "surrounding noise level" such that according to the definition stored in the EEPROM 130 the amplifier 126 applies an additional gain offset to the audio signals sent to the power amplifier 137.
  • the present auditory scene category determined by the classification unit 134 may be characterized by a classification index.
  • the classification unit will analyze the audio signals produced by the microphone arrangement 26 of the transmission unit 102 in the time domain and/or in the frequency domain, i.e. it will analyze at least one of the following: amplitudes, frequency spectra and transient phenomena of the audio signals.
  • Fig. 7 shows schematically the use of an alternative embodiment of a system for hearing assistance, wherein the receiver unit 103 worn by the user 101 does not comprise an electroacoustic output transducer but rather it comprises an audio output which is connected, e.g. by an audio shoe (not shown), to an audio input of a hearing instrument 104, e.g. a hearing aid, comprising a microphone arrangement 36.
  • the hearing aid could be of any type, e.g. BTE (Behind-the-ear), ITE (In-the-ear) or CIC (Completely-in-the-channel).
  • Fig. 8 a block diagram of the receiver unit 103 connected to the hearing instrument 104 is shown.
  • the architecture of the receiver unit 103 of Fig. 8 corresponds to that of Fig. 7.
  • Fig. 9 is a block diagram of an example in which the receiver unit 103 is connected to a high impedance audio input of the hearing instrument 104.
  • the signal processing units of the receiver unit 103 of Fig. 8 are schematically represented by a module 31.
  • the processed audio signals are amplified by the variable gain amplifier 126.
  • the output of the receiver unit 103 is connected to an audio input of the hearing instrument 104 which is separate from the microphone 36 of the hearing instrument 104 (such separate audio input has a high input impedance).
  • the first audio signals provided at the separate audio input of the hearing instrument 104 may undergo pre-amplification in a pre-amplifier 33, while the audio signals produced by the microphone 36 of the hearing instrument 104 may undergo pre-amplification in a preamplifier 37.
  • the hearing instrument 104 further comprises a digital central unit 35 into which the audio signals from the microphone 36 and the audio input are supplied as a mixed audio signal for further audio signal processing and amplification prior to being supplied to the input of the output transducer 38 of the hearing instrument 104.
  • the output transducer 38 serves to stimulate the user's hearing 39 according to the combined audio signals provided by the central unit 35.
  • the receiver unit 103 may control - by controlling the gain applied by the variable gain amplifier 126 — also the ratio of the gain applied to the audio signals from the microphone arrangement 26 and the gain applied to the audio signals from the microphone 36.
  • Fig. 10 shows a modification of the embodiment of Fig. 9, wherein the output of the receiver unit 103 is not provided to a separate high impedance audio input of the hearing instrument 104 but rather is provided to an audio input of the hearing instrument 104 which is connected in parallel to the hearing instrument microphone 36. Also in this case, the audio signals from the remote microphone arrangement 26 and the hearing instrument microphone 36, respectively, are provided as a combined/mixed audio signal to the central unit 35 of the hearing instrument 104.
  • the gain for the audio signals from the receiver unit 103 and the microphone 36, respectively, can be controlled by the receiver unit 103 by accordingly controlling the signal at the audio output of the receiver unit 103 and the output impedance Zl of the audio output of the receiver unit 103, i.e. by controlling the gain applied to the audio signals by the amplifier 126 in the receiver unit 103.
  • the transmission unit to be used with the receiver unit of Fig. 8 corresponds to that shown in Fig. 4.
  • the gain control scheme applied by the classification unit 134 of the transmission unit 102 may correspond to that shown in Fig. 6.
  • the permanently repeated determination of the present auditory scene category and the corresponding setting of the gain allows to automatically optimize the level of the first audio signals and the second audio signals according to the present auditory scene. For example, if the classification unit 134 detects that the speaker 100 is silent, the gain for the audio signals from the remote microphone 26 may be reduced in order to facilitate perception of the sounds in the environment of the hearing instrument 104 - and hence in the environment of the user 101. If, on the other hand, the classification unit 134 detects that the speaker 100 is speaking while significant surrounding noise around the user 101 is present, the gain for the audio signals from the microphone 26 may be increased and/or the gain for the audio signals from the hearing instrument microphone 36 may be reduced in order to facilitate perception of the speaker's voice over the surrounding noise.
  • Attenuation of the audio signals from the hearing instrument microphone 36 is preferable if the surrounding noise level is above a given threshold value (i.e. noisy environment), while increase of the gain of the audio signals from the remote microphone 26 is preferable if the surrounding noise level is below that threshold value (i.e. quiet environment).
  • a given threshold value i.e. noisy environment
  • increase of the gain of the audio signals from the remote microphone 26 is preferable if the surrounding noise level is below that threshold value (i.e. quiet environment).
  • the reason for this strategy is that thereby the listening comfort can be increased.
  • receiver unit 103 and the hearing instrument 104 have been shown as separate devices connected by some kind of plug connection (usually an audio shoe) it is to be understood that the functionality of the receiver unit 103 also could be integrated with the hearing instrument 104, i.e. the receiver unit and the hearing instrument could form a single device.
  • Fig. 11 is a block diagram of a VAD, which is suitable also for applications other than in the transmission unit of the wireless system of Fig. 4, such as in a monaural or binaural hearing instrument system.
  • the audio signals generated by the microphones Ml and M2 of the microphone arrangement 26 may be supplied, after having been digitized in the converters 109 and 1 10, respectively, to a digital signal processor (DSP) 122 via a link 212 and 213, respectively, which may be wired or wireless.
  • DSP digital signal processor
  • one of the links 212, 213 introduces a delay of the transmitted audio signal with regard to the other one of the links 212, 213, a delay compensation will be included in the links 212, 213, usually by delaying the "faster" link accordingly (for example, a wireless link usually involves a signal delay compared to a wired link).
  • the distance between the microphones Ml and M2 of the microphone arrangement 26 may vary from a few mm to 20 cm (the latter corresponds to the ear-to-ear distance).
  • the microphones Ml, M2 may be provided at the same ear, or they may be provided at different ears in order to achieve maximum separation in space for enabling particularly efficient beam forming.
  • the input signals provided via the links 212 and 213 are supplied to a beam- former unit 111 including a beam former implemented by a classical beam former algorithm and a low pass filer, for example, a 5 kHz low pass filter.
  • the audio signals leaving the beam former unit 111 are supplied to an audio signal processing unit 214 which also may include a gain model.
  • the audio signal processing unit 214 also may receive, as additional input, the original input audio signals provided by the links 212 and 213.
  • the output of the beam former unit 1 11 also is supplied to a voice energy estimator unit 114, which is provided for computing the total energy contained in the voice spectrum in the same manner as the unit 1 14 of the embodiment of Fig. 4.
  • the original audio input signals provided by the links 212 and 213 are also supplied to a DOA estimator 219 which determines the DOA value of the input audio signals, for example, by considering the phase difference between the two audio channels.
  • the input audio signals of at least one of the links 212 and 213 are supplied to a surrounding noise level estimator unit 1 17 which produces an output signal supplied to a level definition unit 1 18.
  • the units 117 and 118 correspond to the unit 117 and 118 of the embodiment of Fig. 4.
  • the output signal of the voice energy estimator unit 114, the DOA estimator 219 and the level definition unit 118 are supplied as input to a voice judgement unit 115, which, based on these input signals, decides whether there is a voice source present close to the microphone arrangement 26 or not.
  • the surrounding noise level estimator unit 1 17 is active only if close voice has not been detected.
  • the output of the voice judgement unit 115 is supplied to the audio signal processing unit 214 in order to control the processing of the audio signals in the unit 214 depending on whether close voice has been detected or not.
  • the parameters of the audio signal processing procedure i.e. the audio signal processing mode, can be selected accordingly so that the audio signal processing parameters can be optimized with regard to the presently prevailing auditory scene.
  • the audio signal processing unit 214 may be provided with the output signal of the DOA estimator 219 and the level definition unit 118 in order to more precisely adapt the audio signal processing procedure to the presently prevailing auditory scene.
  • the audio signals processed by the unit 214 may be supplied as audio signals 215 to the stimulating means (typically a loudspeaker) of a hearing instrument.
  • the stimulating means typically a loudspeaker
  • One example of an application of the system of Fig. 11 is a monaural hearing instrument system.
  • the microphones Ml and M2 would be part of the same hearing instrument, and the stimulating means for the audio signals 215 also would be part of the same hearing instrument.
  • FIG. 12 and 13 An example of an application relating to a binaural hearing aid system comprising a right ear hearing aid 302 and a left ear hearing aid 303 worn at the right ear and left ear, respectively, of a user 301 is shown in Figs. 12 and 13.
  • Fig. 12 the use of such a binaural system is schematically shown, with the hearing aids 302 and 303 being separated by the ear-to-ear distance d (which corresponds to about 20 cm) and with the microphone Ml of the right ear hearing aid 302 and the microphone M2 of the left ear hearing aid 303 forming the microphone arrangement 26 of two microphones spaced apart by the distance d.
  • the voice 305 of a speaker 300 is captured both at the microphone Ml and the microphone M2.
  • the hearing aids 302 and 303 are provided with means for establishing a wireless audio signal link 304 between them for exchanging audio signals captured by the microphones Ml and M2.
  • the link 304 may be an inductive link.
  • FIG. 13 a block diagram of the right ear hearing aid 302 is shown.
  • the functionality implemented by the DSP 122 corresponds to that shown in Fig. 1 1, i.e. the units 111, 114, 115, 117, 118, 214 and 219 correspond to that of Fig. 11.
  • the audio signals captured by the microphone Ml are digitized in the converter 109 and undergo a delay compensation in a delay compensation unit 230 prior to being supplied as input to the DSP 122.
  • the audio signals captured by the microphone M2 of the left ear hearing aid 303 are digitized by a converter 110 of the left ear hearing aid 303 and then are transmitted via the wireless audio link 304 to the right ear hearing aid 302 where they are received and, after demodulation, are supplied as input audio signals to the DSP 122.
  • the audio signals captured by the microphone Ml represent one of the audio input channels to the DSP 122 and the audio signals captured by the microphone M2 represent the other audio signal input channel.
  • the delay compensation unit 230 is provided for compensating the delay introduced by the wireless audio link 304, thereby enabling phase analysis of the audio signals provided by the microphones Ml and M2 for beam forming and DOA estimation and for other audio signal processing in the unit 214.
  • the audio signal processing unit 214 which may include a gain model and an auditory scene classifier, may be supplied with the original audio signals from the microphones Ml and M2 and with the output of the beam former unit 11 1. Also the beam former unit is supplied with the audio signals from the microphones Ml and M2 as the input. As in the embodiment shown in Fig. 11, the audio signal processing unit 214 is controlled by the output of the DOA estimator 219, the output of the level definition unit 118 and the output of the voice judgement unit 115.
  • the processed audio signals 215 produced by the unit 214 are supplied to a power audio amplifier 137 and are reproduced by the loudspeaker 136 of the right ear hearing aid 302.
  • the left ear hearing aid 303 has an architecture which is analog to that of the right ear hearing aid 302 shown in Fig. 13, i.e. the left ear hearing aid 303 receives the audio signals captured by the microphone Ml of the right ear hearing aid 302 via the wireless audio signal link 304 and it uses the audio signals captured by the microphone M2 of the left ear hearing aid 302 as direct input.
  • the transmitter for transmitting the audio signals captured by the microphone Ml of the right ear hearing aid 302 via the audio link 304 is shown schematically at 240 in Fig. 13.

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  • Engineering & Computer Science (AREA)
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  • General Health & Medical Sciences (AREA)
  • Neurosurgery (AREA)
  • Otolaryngology (AREA)
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  • Acoustics & Sound (AREA)
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Abstract

L'invention porte sur un procédé permettant d'offrir une assistance auditive à un utilisateur (101, 301), comprenant les étapes consistant à : capturer des signaux audio par un agencement de microphone (26) comprenant au moins deux microphones espacés (M1, M2); estimer l'énergie totale contenue dans le spectre vocal des signaux audio capturés au niveau d'au moins l'un des microphones; estimer la valeur de la direction d'arrivée des signaux audio capturés par comparaison des signaux audio capturés par au moins deux des microphones espacés; déterminer si une voix est présente à proximité de l'agencement de microphone en tenant compte de l'énergie totale estimée contenue dans le spectre vocal des signaux audio capturés et de la valeur estimée de la direction d'arrivée des signaux audio capturés; délivrer un signal représentatif de ladite détermination; traiter lesdits signaux audio capturés selon ledit signal représentatif de ladite détermination; et stimuler l'audition d'un utilisateur, par des moyens de stimulation portés au niveau de ou dans au moins l'une des oreilles de l'utilisateur (39), selon les signaux audio traités.
EP07725080A 2007-05-10 2007-05-10 Procédé et système permettant d'offrir une assistance auditive à un utilisateur Withdrawn EP2165566A1 (fr)

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Families Citing this family (34)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US8811637B2 (en) 2008-12-31 2014-08-19 Starkey Laboratories, Inc. Method and apparatus for detecting user activities from within a hearing assistance device using a vibration sensor
US9473859B2 (en) 2008-12-31 2016-10-18 Starkey Laboratories, Inc. Systems and methods of telecommunication for bilateral hearing instruments
CN102428717B (zh) * 2009-08-11 2016-04-27 贺尔知识产权公司 估计声音波达方向的系统和方法
US8737651B2 (en) 2009-11-17 2014-05-27 Phonak Ag Hearing assistance system and method
US8693715B2 (en) 2009-11-17 2014-04-08 Phonak Ag Hearing assistance system and method
US8515109B2 (en) * 2009-11-19 2013-08-20 Gn Resound A/S Hearing aid with beamforming capability
EP2534854B1 (fr) 2010-02-12 2017-08-09 Sonova AG Système et procédé de transmission sonore sans fil
US20120314890A1 (en) 2010-02-12 2012-12-13 Phonak Ag Wireless hearing assistance system and method
US9137613B2 (en) 2010-02-12 2015-09-15 Phonak Ag Wireless sound transmission system and method
EP2534887A1 (fr) 2010-02-12 2012-12-19 Phonak AG Système et procédé de transmission du son par ondes radioélectriques utilisant un mode de saut de fréquence et d'économie d'énergie amélioré
EP2367286B1 (fr) * 2010-03-12 2013-02-20 Harman Becker Automotive Systems GmbH Correction automatique du niveau de bruit de signaux audio
EP2375781B1 (fr) * 2010-04-07 2013-03-13 Oticon A/S Procédé de contrôle d'un système d'assistance auditive binaurale et système d'assistance auditive binaurale
US9374648B2 (en) 2010-04-22 2016-06-21 Sonova Ag Hearing assistance system and method
EP2561686B1 (fr) 2010-04-22 2017-08-30 Sonova AG Système et procédé d'aide auditive
US8462969B2 (en) * 2010-04-22 2013-06-11 Siemens Audiologische Technik Gmbh Systems and methods for own voice recognition with adaptations for noise robustness
EP2617127B2 (fr) 2010-09-15 2023-10-18 Sonova AG Procédé et système pour fournir à un utilisateur une aide auditive
DK2643983T3 (en) 2010-11-24 2015-01-26 Phonak Ag Hearing assistance system and method
EP2692152B1 (fr) 2011-03-30 2016-07-13 Sonova AG Système et procédé de transmission sonore sans fil
CN103503482A (zh) 2011-05-04 2014-01-08 峰力公司 自学式听力辅助系统及其操作方法
US20130013302A1 (en) 2011-07-08 2013-01-10 Roger Roberts Audio input device
US8989413B2 (en) * 2011-09-14 2015-03-24 Cochlear Limited Sound capture focus adjustment for hearing prosthesis
WO2013107516A1 (fr) 2012-01-20 2013-07-25 Phonak Ag Procédé et transmission de son sans fil
WO2014166525A1 (fr) 2013-04-09 2014-10-16 Phonak Ag Procédé et système pour fournir une aide auditive à un utilisateur
CN105917670A (zh) 2013-12-10 2016-08-31 索诺瓦公司 无线立体声听力辅助系统
WO2016007480A1 (fr) * 2014-07-11 2016-01-14 Analog Devices, Inc. Annulation de bruit de liaison montante de faible puissance
US20160165361A1 (en) * 2014-12-05 2016-06-09 Knowles Electronics, Llc Apparatus and method for digital signal processing with microphones
US10325600B2 (en) * 2015-03-27 2019-06-18 Hewlett-Packard Development Company, L.P. Locating individuals using microphone arrays and voice pattern matching
EP3430821B1 (fr) * 2016-03-17 2022-02-09 Sonova AG Système d'assistance à l'écoute dans un réseau acoustique à locuteurs multiples
US10244333B2 (en) * 2016-06-06 2019-03-26 Starkey Laboratories, Inc. Method and apparatus for improving speech intelligibility in hearing devices using remote microphone
US10621980B2 (en) * 2017-03-21 2020-04-14 Harman International Industries, Inc. Execution of voice commands in a multi-device system
US10396835B2 (en) * 2017-06-16 2019-08-27 Apple Inc. System and method for reducing noise from time division multiplexing of a cellular communications transmitter
CN108810780B (zh) * 2018-06-11 2020-11-24 厦门新声科技有限公司 双耳助听器平衡调节的方法及装置
CN111402916B (zh) * 2020-03-24 2023-08-04 青岛罗博智慧教育技术有限公司 一种语音增强系统、方法及手写板
CN115474117B (zh) * 2022-11-03 2023-01-10 深圳黄鹂智能科技有限公司 基于三麦克风的收音方法和收音装置

Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20040001598A1 (en) * 2002-06-05 2004-01-01 Balan Radu Victor System and method for adaptive multi-sensor arrays

Family Cites Families (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US7254246B2 (en) * 2001-03-13 2007-08-07 Phonak Ag Method for establishing a binaural communication link and binaural hearing devices
US20060182295A1 (en) * 2005-02-11 2006-08-17 Phonak Ag Dynamic hearing assistance system and method therefore

Patent Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20040001598A1 (en) * 2002-06-05 2004-01-01 Balan Radu Victor System and method for adaptive multi-sensor arrays

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US20110044481A1 (en) 2011-02-24
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